I thought I had found an optimal solution in Possio Greta:

https://www.youtube.com/watch?v=ZxZmPJUqrRI
https://www.amazon.com/Possio-Greta-Portable-GSM-Printer/dp/B002SIEM6A?SubscriptionId=AKIAJKHDPWLAVHE4KDDQ
https://gizmodo.com/235790/possio-greta-combination-printer-scanner-fax-and-cellphone

But I talked to the head office in Sweden, and it turned out that it doesn't work everywhere in the world. I can't remember the details now, but I had a lengthy discussion about it, and it's of quite limited use for me when I travel between different parts of the world and different countries.

Otherwise, is there any alternative to smartphone and internet, such as a small portable fax machine or any other device or other type of communication (analogue instead of digital etc.)? Telex, satellite, short-wave radio, SIP for fax etc.?

We have KX-NT700 model, and was working fine with our Panasonic PABX, until we check it today.

The dial tone is missing when we press speaker phone, hence not able to call.

The basic settings are intact as i compared with Phone Manual, such as :
IP + TEL.

We are trying it through the telephone extension line (Ext: 222) , while this line is being used with KX T7665 without any issue.

For your knowledge, this phone had an extension of 300 while it was working fine before.

Please suggest if there is any configuration issue in the phone or this is a mechanical fault in it ?

Thanks in advance.
SYA

My Company just bought the Polycom 7000 and want it hooked up to the Samsung 7400. Documentation for just about any of the Samsung 7400 systems is rare and even harder to find how to set anything up. If anyone has any experience setting up Non-Samsung Phones to Samsung phone systems I would be grateful.

What is the best practice of backing up Cisco Call Manager 11.5 ? (VM)

http://www.samsung.com/uk/smartphones/galaxy-note8/?cid=uk_email_shop_ss-e66-sep2a-sim12085_aid34_20170914_1g-cta-hero-note8&sci=109085541&listid=1370&thcampaignid=12085&thid=6493423E54C0E99C96E57FB11ADD263A&m_i=XWcYEZyTgtxY%2BynmFK4kDQY2iD7Lo25EfeFkqf7NS6Bs80Ryyu5ggPgLdl5TcgIDHwGXDkQs77KdCB5u7AI9Hg9kmwoUX3&jumpto=shop

I'm confused with the above.  Is there just one type of Samsung Note 8 i.e. are they all dual sim compatible?  because the above web site does not explain the different between the two options and on Amazon people are selling the dual sim versions for over GBP 1000 where is the single sim is GBP 800.

Question -
company's IP phones/voice managed by 3rd party.  No PBX on-site.  
Is SIP Trunking still required for the calls or its just calling over the Internet?

I am looking for a free cloud-based call center software such as Fenero. There is nothing wrong with Fenero except that they don't provide any support at all getting started unless you sign up to pay significantly and monthly.

In their comparison at the bottom of this page, http://www.fenero.com/, they mention Five9 and InContact as being more costly solutions. SalesForce is very costly and is out of the question.

We will be making outbound calls and doing emails.

Free or pay-per-call or pay-per-support incident would be a better model getting started than would a monthly option.

Assistance in finding a better solution for this would be greatly appreciated.

For the Current PBX system, the Telco delivered an Adtran, off the Adtran there is an amphenol cable that is punched down to a 66 block.  There is also a CAT5 off the adtran that is also punched down to a 66 block.  

I am installing a Cisco VoIP solution,  but I am not sure how to connect my Cisco voice gateway router with a PRI/T1 card to the Adtran.  I'm not sure if the telco will have to hand the circuit off to me another way, or if I can just connect to the adtran using the ethernet port.  I'm having a lot of trouble getting info from telco.    Could anyone lend me some insight on this and their experiences?   I do have some VOIP experience, but no experience with traditional pbx, 66 blocks, etc.  Thanks.

Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.

SIPp is a free SIP traffic generating tool for Linux.
http://sipp.sourceforge.net/

SIPp user manual says you can install SIPp under CYGWIN on windows. However I am not experienced  with compiling applications to run under Linux and need help getting SIPp up and running under CYGWIN on a windows10 machine.

I have successfully installed CGYWIN and included the following packages (all successfully)
gcc-core
gcc-g++
gcc
libncurses
make

After the CGYWIN install, I put C:/cygwin64/bin in the win10 systems’ environment variable PATH – so far all ok and CGYWIN seems to be working fine.

In addition, the SIPp install instructions state:

Warning
SIPp compiles under CYGWIN on Windows, provided that you installed IPv6 extension for CYGWIN (http://win6.jp/Cygwin/), as well as libncurses and (optionally OpenSSL and WinPcap). SCTP is not currently supported.


QUESTION 1 -  Do you know what this is???    IPv6 extension for CYGWIN http://win6.jp/Cygwin/ 
is it a CYGWIN package, and entire install version??
What/how do I need to do to check/install?

QUESTION 2 – Nothing happens when I try to run “autoreconf -ivf” ...but this might have to do with Question 1 not being addressed yet.

 /cygdrive/c/Backup/tools/SIPp/3.3
$ autoreconf -ivf
-bash: autoreconf: command not found


+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
FROM DOC


Installing SIPp
•      On Linux, SIPp is provided in the form of source code. You will need to…

the system was setup with an auto attendant and 3 call groups for voice mail and after the auto attendant the caller would choose 1 , 2 or 3  and that would send you to the proper mail box and then you would hear that message. i can not figure out how to rerecord the auto attendant message or change the message for the mail boxes that auto attendant send the caller to. i can do intercom 500 from every extension and that takes me to voice mail management, but I can not figure out how to manage another mail box other than the extension I am on or how to change the recording on auto attendant. I can access both systems with my laptop.